System And Method For Automatically Adjusting Floor Controls For A Conversation

ABSTRACT

A system and method for automatically adjusting floor controls for a conversation is provided. Audio streams are received, which each originate from an audio source. Floor controls for a current configuration including at least a portion of the audio streams are maintained. Conversational characteristics shared by two or more of the audio sources are determined. Possible configurations for the audio streams are identified based on the conversational characteristics. An analysis of the current configuration and the possible configurations is performed. A change threshold is applied to the analysis. When the analysis satisfies the change threshold, the floor controls are automatically adjusted. The audio streams are mixed into one or more outputs based on the adjusted floor controls.

CROSS-REFERENCE TO RELATED APPLICATIONS

This patent application is a continuation of U.S. patent applicationSer. No. 10/414,912, filed Apr. 16, 2003, pending, which claims priorityfrom U.S. Provisional Patent Application Ser. No. 60/450,724, filed Feb.28, 2003, expired, the priority filing dates of which are claimed, andthe disclosures of which are incorporated by reference.

INCORPORATION BY REFERENCE

This patent application is related to U.S. patent application Ser. No.10/414,923, filed Apr. 16, 2003, pending, the disclosure of which isincorporated herein by reference.

FIELD

This invention relates to the field of computer-mediated groupcommunication systems, and in particular, to a system and method forautomatically adjusting floor controls for a conversation.

BACKGROUND

Groups of people have communicated together for eons. This communicationincludes styles where a group of people listen to a presenter as well asstyles where people collaborate in a social interaction such as ameeting (among other styles). In the following description, the termmeeting is intended to include all social interactions. Meetings oftenhave subgroups of people who carry on separate conversations within thecontext of the meeting. Each of these subgroups maintains aconversational floor for that subgroup while the members of the subgroupmaintain awareness of the primary group conversation. The primary groupconversation generally continues even though separate conversationalfloors are established. While this conversational style works well whenthe number of participants is small and all of the participants areco-located (such as in a conference room), it is completely disruptiveusing existing technology that enables remote parties to communicatewith each other (for example, teleconference technology, two-way sharedradio channels, etc.).

An example of this problem is that of a “party line” telephone orteleconference call in which there is no capability to schism theconversation into separate conversational floors. This is also true ofshared-channel radio systems such as police and fire-fighter radiocommunications. Communications between the participants are mixedtogether on the communication channel making it difficult for arbitraryusers to communicate and often requiring complex protocols among theusers to provide some order to the communications. Although somehigh-end teleconference systems support multiple conversational floors(for example, by “sub-conferencing” or by channel switching) theestablishment and modification of these conversational floors isdifficult. This difficulty lessens the spontaneity of establishing aconversational floor in a remote meeting.

Instant messaging and chat systems allow for schisming as a number ofusers can participate in a textual “chat room” where each user's typedmessage is displayed to all the members of the room (subject toper-member controls). Each user can also explicitly create and maintaina side-channel textual conversation with a subgroup of members. Thisschisming is not automatic but requires that explicit user commands bemade to the system.

U.S. Pat. No. 6,327,567 B1 to Willehadson et al., entitled Method andSystem for Providing Spatialized Audio in Conference Calls, and filedFeb. 10, 1999 teaches a mechanism that allows sub-conferences(conversational floors). However, each user needs to manually enter adialog with a command unit (by DTMF, by a user interface with a controlunit or by a recognized voice command) to initiate or participate in asub-conference or to switch between sub-conferences. In addition,Willehadson uses a complicated spatialization technique to indicate whatsub-conferences are available. Willehadson does not teach automaticdetection of conversational floors or automatic switching betweenconversational floors responsive to the conversational characteristicsrelated to the conversations.

It would be advantageous to provide a capability that addresses theabove-mentioned problems.

SUMMARY

One aspect of the invention is a method of identifying a conversation.This method includes the step of extracting streams of feature data froma conversation communication between users over a communicativeinterval. The streams of feature data are then analyzed in variouscombinations of users over a communicative interval to identify aconversation between two or more of the users. Other aspects of theinvention include apparatus for implementing the method, and computerprogram products for causing a computer to implement the method.

Another aspect of the invention is a method for an audio processingsystem that receives one or more audio streams where each audio streamis from one of a plurality of audio sources. A distinguishing step thendistinguishes one or more audio substreams from the audio streams. Amixing step mixes the substreams (responsive to a plurality of floorcontrols) for one or more outputs that are associated with the audiosources. An analyzing step analyzes (for a plurality of users associatedwith the plurality of audio sources) one or more conversationalcharacteristics of two or more of the plurality of users. Another stepautomatically adjusts the floor controls responsive to the step ofanalyzing. Other aspects of the invention include apparatus forimplementing the method, and computer program products for causing acomputer to implement the method.

A further aspect provides a system and method for automaticallyadjusting floor controls for a conversation is provided. Audio streamsare received, which each originate from an audio source. Floor controlsfor a current configuration including at least a portion of the audiostreams are maintained. Conversational characteristics shared by two ormore of the audio sources are determined. Possible configurations forthe audio streams are identified based on the conversationalcharacteristics. An analysis of the current configuration and thepossible configurations is performed. A change threshold is applied tothe analysis. When the analysis satisfies the change threshold, thefloor controls are automatically adjusted. The audio streams are mixedinto one or more outputs based on the adjusted floor controls.

The foregoing and many other aspects of the present invention will nodoubt become obvious to those of ordinary skill in the art after havingread the following detailed description of the preferred embodimentsthat are illustrated in the various drawing figures.

BRIEF DESCRIPTION OF THE DRAWINGS

The embodiments of the invention will be described in detail, withreference to the following figures wherein:

FIG. 1 illustrates an architecture in accordance with an embodiment;

FIG. 2 illustrates a group communication server in accordance with anembodiment;

FIG. 3 illustrates a group communication server architecture inaccordance with an embodiment;

FIG. 4 illustrates a remote digital audio microphone system inaccordance with an embodiment;

FIG. 5 illustrates an embodiment of a floor control data structure inaccordance with a embodiment;

FIG. 6 illustrates a group communication server initialization thread inaccordance with an embodiment;

FIG. 7 illustrates a floor configuration thread in accordance with anembodiment;

FIG. 8 illustrates an audio source handler thread in accordance with anembodiment;

FIG. 9 illustrates a new active source active thread in accordance withan embodiment;

FIG. 10 illustrates a pre-analysis thread in accordance with anembodiment;

FIG. 11 illustrates a user output thread in accordance with anembodiment;

FIG. 12 illustrates a configuration generation thread in accordance withan embodiment;

FIG. 13 illustrates a turn-taking analysis thread in accordance with anembodiment;

FIG. 14 illustrates a referential-action analysis initialization threadin accordance with a embodiment;

FIG. 15 illustrates one embodiment of a referential-action analysisthread in accordance with a embodiment;

FIG. 16 illustrates one embodiment of a responsive-action analysisinitialization process in accordance with a embodiment;

FIG. 17 illustrates one embodiment of a responsive-action analysisthread in accordance with a embodiment; and

FIG. 18 illustrates example graphs that can be useful in understandingthe threads of FIG. 7 and FIG. 12.

DETAILED DESCRIPTION

One aspect of the invention is a media communication system employingautomatic detection of human conversational behavior. In one embodiment,the system makes probabilistic inferences of conversational groupmembership based on conversational characteristics such as quantitativetemporal relationships between specific audible cues and actions duringconversation (“temporal features”). One embodiment of the system neednot require an understanding of human speech as it uses the ability todetect the presence or absence of a given user's vocalizations and/or todetect specific vocalizations within the speech with a reasonable degreeof reliability.

The conversational characteristics can include both audiocharacteristics and physiological characteristics (such as can bedetermined by a biometric device). Although much of the followingexplicitly describes the analysis of audio characteristics, one skilledin the art can, after reading the disclosure herein, apply equivalenttechniques to any available physiological characteristics that can bereceived by the group communication server 103.

The detection of the presence/absence of a user's vocalizations can bedone by capturing audio with a throat microphone or a “bone-conduction”microphone and then using a simple signal detection algorithm (e.g., oneusing energy thresholds and/or signal zero-crossing rates) to segmentthe audio into speech/silence. (A conventional microphone may be used tocapture the user's speech for human listening purposes—the use ofspecial microphones for speech detection is beneficial but not necessarybecause it decreases the amount of external noise for which the signaldetection algorithm must allow.)

The temporal features can be classified using known pattern recognitiontechniques. One way to do this is to compare quantitative feature valuesto known distributions (e.g., the audio from a multiparty conversationcan be segmented, hand-labeled and used to train a Naïve Bayesclassifier). If a given relationship is known to be universal acrossusers (or nearly so), a static distribution can be used. If a givenrelationship is subject to variation across users, an initialdistribution can be incrementally modified. Some temporal featuresgenerally indicate membership in the same conversational floor. Speakersin the same conversational floor place the beginning of theirvocalizations at transition relevance places (TRPs), temporally aligningthem with pauses in other speakers' turns. This general principal can beused to develop a quantitative feature.

One example feature computes the time difference between the currentstart endpoint of speaker X's turn, t, and the previous final endpointof speaker Y's most recent turn prior to t. This difference can becompared to a decreasing distribution (e.g., the comb-shapeddistribution of Wilson & Zimmerman (Wilson, T. P. and Zimmerman, D. H.,The Structure of Silence Between Turns in Two-Party Conversation,Discourse Processes 9 (1986), 375-390) or an empirical distribution) toproduce an estimated probability that the speaker X and speaker Y arecommunicating.

Another example feature uses the characteristic that speakers do notsimply wait for and react to the completion of other speakers' turns;rather, they project (i.e., anticipate) a turn's completion based oncues such as content and prosody. Since speaker X may misjudge when theprevious speaker Y will finish a vocalization, the starting endpoint ofX's turn sometimes starts before the final endpoint of Y's turn. Thisfeature computes the time difference between the current start endpointof speaker X's turn, t, and the final endpoint of speaker Y's turn thatis closest to t (which may therefore be before or after t). Thisdistribution can also be modeled empirically; again, longer differencesare less likely to represent a deliberate alignment and thus, it is lesslikely that the speakers are in the same conversational floor.

Yet another example feature can look for evidence that the speakers arenot in the same conversational floor. For example, while speakers dooften overlap their speech (as when speakers start at the same time, orwhen speakers complete a sentence together), it is very unusual to seesustained periods of speech that overlaps with other speakers' speech ifthe speakers in question are in the same conversational floor. Thisgeneral principal can be used to develop a quantitative feature.

One example algorithm is to determine a sliding time window T, andwithin T compute a vector corresponding to periods of simultaneousspeech given some time quantization (for example, 1 millisecond). Thencompute the scalar product of this vector with a weighting vector. Theresulting value can be compared to an empirical distribution todetermine a quantitative measurement for the feature.

Once the features are determined, they can then be used to computelikelihoods, posterior probabilities, and conversational floorconfigurations as is subsequently described.

Additional evidence of membership in a conversational floor can beobtained by recognizing particular vocalizations (for example, user orgroup names, or backchannel acknowledgement vocalizations). Detailsabout these techniques are also subsequently described.

Physiological characteristics can also be compared with the user's audioto determine a feature.

One skilled in the art after reading the following will understand thata user can have a microphone or other audio pickup, and a speaker. Theuser's audio pickup generates signals that result in digitized packetsthat are identified as to their source (the source being, for example,the user). Digitized audio packets for output to the user are similarlyidentified, and these packets are generally sent back to the user forreproduction through the user's speakers or headphone. One aspect of theinvention acquires audio from multiple sources, mixes the source's audioresponsive to which user/source the mixed audio will be delivered, andautomatically adjusts the mix responsive to an analysis of theconversational characteristics such as the vocalizations of the users.Such a one will also understand that streams of feature data can beextracted from the conversational communication between people. Theconversational communication can be comprised of textual information,audible information, visual information, tactile information or somecombination thereof. The conversational communication can be manually orautomatically transcribed.

FIG. 1 illustrates an architecture 100 that includes a network 101, agroup communication server 103 and a plurality of audible sources 105.The network 101 can be a LAN, WAN, the Internet, or any other networksuitable for transporting audio information whether in digital or analogform. The plurality of audible sources 105 can include microphones thatreceive a person's vocalizations or vocalizations (and can also includea mechanism for detecting the occurrence of a person's vocalization tobetter separate vocalization from background noise—for example, a throatmicrophone). In addition, any of the plurality of audible sources 105can be connected to the network 101 using wire or wireless technologies.Further, at least some of the plurality of audible sources 105 have somemechanism for recording or presenting audio information (for example, byproviding signals capable of driving an audio reproduction mechanismsuch as an earphone, a hearing aid, a bone conduction transducer, adirect tympanic stimulator, a headphone set, a speaker, etc.). Oneskilled in the art would be able to create a similar architecturedirected to textual processing or other group communication processingwithout undue experimentation after having read the disclosure herein.

Any of the plurality of audible sources 105 can include signalprocessing capability for converting analog audio information intodigital audio information and for sending the audio information to thegroup communication server 103. These capabilities can be included aspart of a microphone, a headset, provided by a portable audio processor,or provided by a server processor in wire or wireless communication withthe microphone.

The group communication server 103 can be a component of a radio system,a wired, wireless and/or cellular telephone system, a component in anemergency control center, a server for an interne audio-based chat room,or a component in any individually addressable group audio system.

FIG. 2 illustrates a group communication server 200 that incorporatesthe invention. The group communication server 200 includes a computer201 that incorporates a CPU 203, a memory 205, and a network interface207. The network interface 207 provides the computer 201 with access toa network 209. The computer 201 also includes an I/O interface 211 thatcan be connected to a user interface device(s) 213, a storage system215, and a removable data device 217. The removable data device 217 canread a computer readable media 219 that typically contains a programproduct 221. The storage system 215 (along with the removable datadevice 217) and the computer readable media 219 comprise a file storagemechanism. The program product 221 on the computer readable media 219 isgenerally read into the memory 205 as a program 223. In addition, theprogram product 221 can be provided from the network (generally encodedwithin an electromagnetic carrier wave—including light, radio, andelectronic signaling) through the network interface 207.

One skilled in the art will understand that not all of the displayedfeatures of the computer 201 need to be present for the invention. Oneskilled in the art will understand that the network transmitsinformation (such as data that defines audible information as well asdata that defines a computer program). Generally, the information isembodied within a carrier-wave. The term “carrier-wave” includeselectromagnetic signals, visible or invisible light pulses, signals on adata bus, or signals transmitted over any wire, wireless, or opticalfiber technology that allows information to be transmitted over anetwork. Programs and data are commonly read from both tangible physicalmedia (such as a compact, floppy, or magnetic disk) and from a network.Thus, the network, like a tangible physical media, is a computer usabledata carrier.

FIG. 3 illustrates a group communication system architecture 300 thatillustrates one embodiment of the invention. This architecture can beimplemented in circuitry, in a suitably programmed computer, or using acombination of programmed computer and circuitry. A user is associatedwith a source and an output. A user's vocalizations are provided to anaudio source and the vocalizations of every ‘other user’ (possiblymodified) are mixed and presented to an output associated with the userfor presentation to the user. Both the sources and the outputs can be“virtual” in that the result is a digitized audio packet that includesan identification (for example, the identification can be that of auser, a virtual bi-directional channel, or include separate identifiersfor the source and output virtual channels) and a timestamp. Thetimestamp and the source/output identification can be added to thepacket by a tagger.

The group communication system architecture 300 includes an audio mixer301 that can be conditioned to mix digital audio from N sources to N ormore outputs where the audio from each source can be individuallycontrolled when mixed for each of the outputs. The audio mixer 301 canthus mix the various sources specifically for each output. This mixingcan be controlled by a set of mixing parameters or floor controls.

Generally, there is an audio source and an output dedicated to each user(although outputs that are not associated with a source are contemplatedby the inventors for monitoring purposes). In addition, additionalinputs can be included that do not have a corresponding output (forexample, computer generated system status messages, or overridecommunications).

The group communication system architecture 300 also includes a flooranalysis module 303 that is used to analyze data for conversationalcharacteristics. The results of the floor analysis module 303 can beused to automatically determine establishment of conversational groupmembership. The floor analysis module 303 then effectuates theconversational floors by specifying the set of mixing parameters for theaudio mixer 301. The set of mixing parameters are communicated from thefloor analysis module 303 to the audio mixer 301 over a floor controlpath 305.

The group communication system architecture 300 includes a first digitalaudio source and tagger 307 through an n^(th) digital audio source andtagger 309 and may include an analog source 311 that feeds an audiodigitizer and tagger 313. The tagger portion of these elements insertssource identification and can insert time-stamp information into thedigitized audio packets. These audio sources can be receivers thatreceive the audio streams.

The digitized and tagged audio data is provided to the inputs of theaudio mixer 301 over one or more mixer audio data paths 315. The digitalaudio data is also provided, via one or more analysis audio data paths317 to the floor analysis module 303. The floor analysis module 303analyzes the available conversational characteristics (for example, ofthe audio data received over the one or more analysis audio data paths317) to determine the values of the set of mixing parameters.

The audio mixer 301 communicates the mixed output audio over one or moreoutput audio data paths 319 to a first digital audio output 321 throughan n^(th) digital audio output 323 and optionally over an audio digitalto analog converter (D/A) 325 connected to an analog audio output 327.

One skilled in the art will understand that, although multiple sourcesand output are indicated in FIG. 3, once the audio packets are digitizedand contain a source identifier, a single physical input/outputcommunication mechanism (for example, a network interface) is sufficientto transmit or receive the packets. The relevant requirement is that theaudio from each source is identified. That identification can beaccomplished by a digitizing input mechanism, by electronics that is incommunication with the digitizing input mechanism, or by electronicsthat is associated with a particular physical source input or othermechanisms well understood in the art. Furthermore, the digitized audiofrom a given source can be associated with a virtual device thatprovides an interface to the audio stream from that source. The separatepaths indicated in FIG. 3 for the one or more mixer audio data paths315, the one or more analysis audio data paths 317, and the one or moreoutput audio data paths 319 can be a single path carrying digitizedaudio packets that are identified according to source. Such a one willalso understand that there are a multitude of techniques for handlingaudio data and for temporally aligning the audio data each of which arecontemplated for use by the inventors. Furthermore, the techniquesrelated to temporally aligning the audio data can be performed by eachof the plurality of audible sources 105, can be done at the groupcommunication server 103 or in some combination.

An ‘additional response analysis’ module 328 can be included thatreceives non-audio data such as video information, biometricinformation, eye-tracking information etc. The ‘additional responseanalysis’ module 328 can analyze this data to determine conversationalcharacteristics that can be correlated with the audio informationprovided by the one or more analysis audio data paths 317 as well asinformation developed by other modules in the floor analysis module 303.The analysis results of the ‘additional response analysis’ module 328are incorporated with the other analysis performed by the floor analysismodule 303 to control the audio mixer 301.

The floor analysis module 303 also analyzes conversationalcharacteristics and can also contain one or more analysis modules suchas a ‘turn-taking analysis’ module 329, a ‘responsive action analysis’module 331, and/or a ‘referential action analysis’ module 333 as well asa conversational floor configuration thread as will be subsequentlydescribed primarily with respect to FIG. 7, FIG. 13, FIG. 15, and FIG.17. The results of the analysis is used to control the floor controlsand/or set of mixing parameters used by the audio mixer 301.

The first digital audio source and tagger 307 through the n^(th) digitalaudio source and tagger 309 and the ID and time stamp portion of theaudio digitizer and tagger 313 are used to identify the input source ofthe digitized data and can mark the data with the time it was received.

One skilled in the art will understand that other embodiments of theinvention can use hybrid digital/analog circuitry for the audio mixer301. In such an embodiment, the floor analysis module 303 could provideits own audio digitizers. Such a one will also understand that the firstdigital audio source and tagger 307 through the n^(th) digital audiosource and tagger 309 and the analog source 311 are generally associatedwith the corresponding output. Each source/output is usually associatedwith a user who performs vocalizations that are picked up by an inputdevice (such as a microphone) and digitized. The user also listens tothe audio produced by the corresponding audio output.

The audio information from each source/output to and from the groupcommunication server 103 can be made available over one or more portsinto the group communication server 103 but generally the digitizedaudio information is packetized and communicated over a network. Inaddition, there need not be a one-to-one correspondence between thesources and the outputs (thus allowing for monitoring an output, or foran overriding source).

FIG. 4 illustrates a remote digital audio microphone system 400 that canbe used to provide digitized audio data to any of the first digitalaudio source and tagger 307 through the n^(th) digital audio source andtagger 309. The remote digital audio microphone system 400 includes anaudio input 401, can include a voicing detector 403 to determine wheninput from the microphone should be processed, and a digitizer 405 thatdigitizes analog signals from the audio input 401 (possibly responsiveto the voicing detector 403). In addition, the remote digital audiomicrophone system 400 includes a communications interface 407 that sendsand receives information to the group communication server 103 over thenetwork 101. In addition, the remote digital audio microphone system 400can include an optional side tone generator 409 (that feeds some of theaudio received by the audio input 401 to the audio output 413) andincludes a D/A converter 411 for converting digital audio informationreceived by the communications interface 407 into analog informationthat can be presented by an audio output 413 (such as a headphone orspeaker). One skilled in the art will understand that FIG. 4 representspossible features in the remote digital audio microphone system 400 andthat these features can be combined in many different ways. Furthermore,the arrangement of devices in FIG. 4 need not imply the order thatsignals are processed.

One skilled in the art will understand that the remote digital audiomicrophone system 400 need not be as fully featured as shown. It onlyneed include the audio input 401, the audio output 413, and some meansof communicating the audio information to and from the groupcommunication server 103.

FIG. 5 illustrates a floor control data structure 500 that includes asource data structure 501 that has a number-of-sources field 503 and asource/output-structure-pointer array 505. The number-of-sources field503 specifies the number of active sources (in some implementations thisfield can specify the total number of possible sources). Thesource/output-structure-pointer array 505 associates each source with apointer to an output data structure 507. The output data structure 507includes a number of outputs field 509 that specifies the number ofoutputs to which audio from the source is to be distributed. Thecontents of the number-of-sources field 503 and the output datastructure 507 is generally expected to be the same (but can be differentfor transitional states or for special purpose applications). The outputdata structure 507 also includes an output control array 511 that caninclude an output identification field, a volume field and an audiomodification field.

The source data structure 501 is used to distribute audio informationfrom the associated source to each output as modified by the volume andaudio modification fields for that output. The volume and audiomodification fields are automatically adjusted by the floor analysismodule 303 through the floor control path 305. Thus for each output, theaudio mixer 301 can mix contributions from each source as specified bythe floor control data structure 500.

The audio modification field is used to apply special effects to themixed audio to help the user better distinguish audio originating from adifferent conversational floor as compared to audio originating from theuser's current conversational floor. These special effects for the floorindication mechanism can include adding echo, intentional masking and/orother effects. This field can also be used to override theconversational floors for a communication (for example, for an overridecommunication that must be heard by all). Furthermore the audiomodification field (or an additional field) can include spatializationcontrols to change the apparent audio position of the audio source.Thus, in one embodiment, the audio from members of ‘this user's’ floorcan always be auditorally placed in front of ‘this user’ while audiofrom other floors is placed to the side or behind ‘this user’.

One skilled in the art, after reading the disclosure herein, can expandthe floor control data structure 500 to support video focusing in avideo conference situation. That is, as a conversational floor isestablished between participants of a video conferencing session, themembers of the conversational floor have a different videorepresentation than do the members of another conversational floor.These modifications can result in highlighting the views containing the‘other users’ on ‘this user's’ conversational floor. The highlightingcan be done by size, color, placement, etc. of the view. Furthermore,such a one will understand that in the textual communicationenvironment, that the output control array 511 can contain fields thatcontrol the appearance of the text of the communication (for example,size, color, shading etc.) to indicate floor membership. Both thetextual and video representations are examples of visual indications offloor membership. In addition, other visual representations of the floorconfiguration can be presented to the user(s) by graphical, textual, orother indicator mechanisms.

Describing now an embodiment of the group communication systemarchitecture 300. This embodiment is described in the context of amulti-threaded procedural programming environment for use by ageneral-purpose computer. One skilled in the art will understand thatthere are many programming methodologies that can be used to program ageneral-purpose computer system in accordance to the group communicationsystem architecture 300. In addition, such a one would understand thatsignificant portions (or totality) of the group communication systemarchitecture 300 can be implemented using special purpose circuitry.

FIG. 6 illustrates a ‘group communication server initialization’ thread600 that is invoked as the group communication server 103 prepares toperform its function. The ‘group communication server initialization’thread 600 initiates at a ‘start’ terminal 601 and continues to an‘initialization’ procedure 603 that performs any requiredinitializations (such as establishing appropriate data structures,enabling communications with the plurality of audible sources 105, andother initializations that would be apparent to one skilled in the art).After the ‘initialization’ procedure 603 completes, the ‘groupcommunication server initialization’ thread 600 continues to an ‘invokefloor configuration thread’ procedure 605 that invokes the threadsubsequently described with respect to FIG. 7 and an ‘invoke sourcethread’ procedure 607 that invokes the thread subsequently describedwith respect to FIG. 8. The ‘group communication server initialization’thread 600 then completes through the ‘end’ terminal 609.

To automatically configure the audio mixer 301, each user'svocalizations are analyzed (as is subsequently described) and theresults of the analysis are used to automatically modify the floorcontrol data structure 500. In one embodiment, this analysis isperformed by one or more threads-of-execution.

A procedure is a self-consistent sequence of computerized steps thatlead to a desired result. These steps can be defined by one or morecomputer instructions. These steps can be performed by a computerexecuting the instructions that define the steps. Thus, the term“procedure” can refer (for example, but without limitation) to asequence of instructions, a sequence of instructions organized within aprogrammed-procedure or programmed-function, or a sequence ofinstructions organized within programmed-processes executing in one ormore computers. Such a procedure can also be implemented directly incircuitry that performs a function that is equivalent to the functionperformed by a computer executing the instructions.

FIG. 7 illustrates a ‘floor configuration’ thread 700 invoked by the‘invoke floor configuration thread’ procedure 605 of FIG. 6 and thatinitiates at a ‘start’ terminal 701. This thread is responsible forreceiving the results of the individual analysis modules, automaticallydetermining and selecting the most likely configuration of audio sourcesfrom these results (thus determining conversational group membership),and for configuring the audio mixer 301 by adjusting values in the floorcontrol data structure 500.

An ‘initialization’ procedure 703 provides any required initializationincluding the possible allocation and initialization of the floorcontrol data structure 500 (if needed), initialization of the floorcontrol path 305, the one or more mixer audio data paths 315, the one ormore analysis audio data paths 317, and other initializations as wouldbecome apparent to one skilled in the art. After the ‘floorconfiguration’ thread 700 completes its initialization, it continues toa ‘receive analysis results’ procedure 705 that receives results fromseparate analysis threads such as those subsequently described withrespect to FIG. 13, FIG. 15, and FIG. 17. Once a result from an analysismodule is received, a ‘determine configuration probabilities’ procedure707 determines the probability for at least some of the possibleconfigurations that that configuration matches how the users areinteracting in each conversational floor.

The per-configuration probabilities can be determined using a variety ofmethods. One method is that of maintaining a participant graph with anode for each user and with an edge between that user and each ‘otheruser’. Each edge in the participant graph is weighted with the pair-wiseprobability that the two users connected by the edge are inconversation. The configuration with the highest mean weight is the mostprobable configuration. This example is subsequently described withrespect to FIG. 18.

One skilled in the art will understand that as the number of sourcesincrease, that the number of possible configuration combinations becomesmuch more numerous. While the probability for each possibleconfiguration can be determined for small numbers of sources, as thenumber of sources increase the number of configurations can be managedusing techniques known in the art.

Once the probabilities are determined for the relevant configurations, a‘select most probable configuration’ procedure 709 selects the currentmost probable configuration. Then, a ‘configure floors’ procedure 711analyzes the past history of the selected configurations and, ifappropriate, will reconfigure the set of mixing parameters in the floorcontrol data structure 500 (thus, changing the conversational floorconfiguration).

The ‘configure floors’ procedure 711 can also apply some hysteresis-likeeffects so that the conversational floor configuration does not changetoo rapidly (which results in a fluttering effect). Thus, it is usefulto maintain a configuration for a minimum number of timeslices. Avariety of methods can be used to determine this. For example, oneoption is to require a single configuration be maintained for aspecified number of consecutive timeslices, another approach is torequire a “random walk” distance of a specified number of timeslicesbefore a change in the selected configuration is allowed (for example,if configuration A is the currently selected configuration,configurations B, C or D must be chosen a net total of 30 moretimeslices relative to A before a configuration other than A can beselected). Other heuristic analysis can also be performed to help keepthe conversational floors stable without interfering with the abilityfor the users in the conversational environment to have sideconversations or otherwise switch conversational floors. The ‘configurefloors’ procedure 711 changes the values in the floor control datastructure 500 to change the conversational floor configuration. Afterthe ‘configure floors’ procedure 711, the ‘floor configuration’ thread700 continues back to the ‘receive analysis results’ procedure 705 toreceive and process more analysis results.

Additional methods can be used to control the ‘configure floors’procedure 711, in accordance with a state-based conversational model.For example, transitions between floor configurations can be controlledusing deterministic state machines. One way to do so is to usedeterministic finite state machines in which individual states representparticular floor configurations, but it is also possible for multiplestates to correspond to particular floor configurations and vice versa.Alternatively, transitions between floor configurations can becontrolled using probabilistic state machines, such as those based onMarkov models or Hidden Markov Models (HMMs).

It should also be apparent that the ‘configure floors’ procedure 711need not perform all types of analysis at the same temporal granularityused in the rest of the system. For example, even if audio samples areprocessed using a fixed 30 ms time granularity, the selection processfor floor configuration can use features corresponding to variable-sizedtime units. One such approach uses so-called “segment-based” analysis,in which features are combined with an explicit time label indicatingthe duration for which the feature value is valid (segment length).

Users can be provided with tools to specify parameters that influencethe operation of the ‘floor configuration’ thread 700. For example,particular preferences can be provided to specified users such that theyare members of all conversational floors. In addition, vocalizationsfrom some designated members (override members) may be given an overridecharacteristic such that vocalizations from the override member causeother member's vocalizations to be silenced for the duration of theoverride member's communication. Another example is the provision of a“hold” mechanism that allows the user to require the ‘floorconfiguration’ thread 700 to maintain its current floor configurationwith respect to that user. This “hold” mechanism and other usefulmechanisms to override the automatic floor control can be invoked bytraditional explicit user commands either vocal or by explicit useraction. Further, in some embodiments, the system can be forced into amanual mode where the floor membership is explicitly controlled by theusers.

FIG. 8 illustrates an ‘audio source handler’ thread 800 invoked by the‘invoke source thread’ procedure 607 of FIG. 6 and that initiates at a‘start’ terminal 801 and continues to an ‘initialization’ procedure 802that performs any one-time initialization for the thread. Then the‘audio source handler’ thread 800 continues to a ‘receive packet’procedure 803 that receives a packet of digitized audio information fromany of the one or more mixer audio data paths 315. A ‘newidentification’ decision procedure 805 determines whether the packet wasfrom a new source (by examining the packet identification informationinserted in the packet by the ID and time stamp portion of the sourceinput). If a new source is detected, the ‘audio source handler’ thread800 continues to an ‘invoke new active source thread’ procedure 807 thatstarts a thread (as is subsequently described with respect to FIG. 9)for the newly activated source.

Otherwise, if the packet identifies its source as one that has beenpreviously active, the ‘audio source handler’ thread 800 continues to a‘distribute packet to source pre-analysis thread’ procedure 809 thatsends a copy of the packet to the pre-analysis thread that is describedwith respect to FIG. 10. An ‘index into output table’ procedure 811accesses the floor control data structure 500 to access the outputcontrol array 511 for audio information from the identified source. An‘iterate each output’ procedure 813 then iterates each output in theoutput control array 511.

For each iterated output, an ‘adjust packet WRT output parameters’procedure 815 creates a copy of the packet as adjusted with respect tothe contents of the volume field and the modification code field. Next,a ‘distribute packet to output thread’ procedure 817 distributes theadjusted packet to the appropriate output thread. The output thread issubsequently described with respect to FIG. 11. After adjusted copies ofthe source packet have been distributed to the appropriate outputthreads (generally all of the output threads, but if the volume fieldindicates that the output is to have no contribution from thisparticular source, the packet need not be distributed to that output),the source packet can be released.

One skilled in the art will understand that the packet adjustment can bedone by the output thread instead of by the ‘audio source handler’thread 800.

Once the ‘iterate each output’ procedure 813 completes the iteration,the ‘audio source handler’ thread 800 continues back to the ‘receivepacket’ procedure 803 to receive the next audio packet.

One skilled in the art will understand that in another embodiment copiesof the source packet can be first distributed to the output queues, andthen the original packet can be sent to the ‘distribute packet to sourcepre-analysis thread’ procedure 809. Both approaches provide theequivalent functionality, but one may be more conducive toparallelization.

It should be noted that the described implementation will drop the firstaudio packet from a newly activated source. This is a one-timeoccurrence. One skilled in the art after having read this descriptioncould re-flow the initial packet back onto the queue because the packetis already time-stamped.

FIG. 9 illustrates a ‘new active source’ thread 900 that is invoked bythe ‘invoke new active source thread’ procedure 807 of FIG. 8. The ‘newactive source’ thread 900 is used to initialize the threads for handlinga new user, for modifying the participant graph, and for initializingthe analysis thread needed for the new user.

The ‘new active source’ thread 900 initiates at a ‘start’ terminal 901.An ‘initialization’ procedure 902 performs any required initialization.An ‘add output table entry’ procedure 903 updates the floor control datastructure 500 to reflect the existence of the newly active source. Oncethe floor control data structure 500 is updated, the ‘new active source’thread 900 can invoke the user output thread (that is subsequentlydescribed with respect to FIG. 11) through an ‘invoke output thread’procedure 905. Once the user output thread starts execution, the newuser will start to hear audio responsive to the initialization performedwhen the ‘add output table entry’ procedure 903 initialized the newentry in the floor control data structure 500.

An ‘invoke source pre-analysis thread’ procedure 907 invokes thepre-analysis thread that is subsequently described with respect to FIG.10. An ‘add source to participant graph’ procedure 909 adds the newlyactivated source to the participant graph (as a new user) and an ‘invokeconfiguration generation thread’ procedure 911 invokes the configurationgeneration thread that is subsequently described with respect to FIG.12.

An ‘invoke turn-taking analysis thread’ procedure 913 then starts theturn-taking analysis thread that is subsequently described with respectto FIG. 13.

An ‘initialize referential analysis’ procedure 915 and an ‘initializeresponsive analysis’ procedure 919 then invoke threads for initializingthe referential analysis and responsive analysis threads as aresubsequently described with respect to FIG. 14 and FIG. 16 respectively.

One skilled in the art will understand that other analysis threads,configuration threads, or capability threads can also be invoked for thenew user at this time.

Finally, the ‘new active source’ thread 900 completes through an ‘end’terminal 923. At this point, the audio mixer 301 is conditioned to addthe audio received by the newly enabled source to the mix generated foreach of the outputs. Thus, the user associated with the audio sourcewill be able to hear the audio corresponding to the default mix. Theconfiguration generation thread has been activated, and once theconfigurations that incorporate the new source are generated, and theanalysis modules initialized, the floor analysis module 303 will be ableto adjust the mix for the new user.

FIG. 10 illustrates a ‘pre-analysis’ thread 1000 that can be used toaccumulate a timeslice of the received audio data and to perform ananalysis of when the audio data represents a vocalization. To summarize,digital audio packets are received (sent by the ‘distribute packet tosource pre-analysis thread’ procedure 809 of FIG. 8) and accumulated ina source-specific buffer until a timeslice of about 30 milliseconds ofaudio is captured. Once the timeslice is captured, it is analyzed todetect periods of vocalization (by a voice activity detector (VAD)). TheVAD analysis generates a bit vector that represents whether vocalizationis detected for the millisecond corresponding to the bit position in thevector. Each VAD vector is added to a VAD buffer for use by the analysisroutines.

The ‘pre-analysis’ thread 1000 is invoked by the ‘invoke sourcepre-analysis thread’ procedure 907 of FIG. 9, initiates at a ‘start’terminal 1001, and continues to an ‘initialization’ procedure 1003 toperform any required initialization. A ‘receive source packet’ procedure1005 receives the audio packet from the ‘distribute packet to sourcepre-analysis thread’ procedure 809. Next, an ‘add packet to sourcebuffer’ procedure 1007 adds the received audio packet to the timeslicedata. A ‘timeslice complete’ decision procedure 1009 determines whetherthe timeslice buffer is full, and if the timeslice buffer is not full,the ‘pre-analysis’ thread 1000 continues back to the ‘receive sourcepacket’ procedure 1005 to accept the next packet.

However, if the ‘timeslice complete’ decision procedure 1009 determinesthat the timeslice is complete, the ‘pre-analysis’ thread 1000 continuesto a ‘VAD timeslice’ procedure 1011 that applies a voice activitydetector (VAD) to the data in the timeslice to determine what portionsof the timeslice correspond to vocalization. The ‘VAD timeslice’procedure 1011 can also use information from a vocalization detectionmechanism (for example, using a signal derived from a throatmicrophone). Once the VAD analysis completes for the timeslice, theresults of the analysis are added to a VAD buffer. An ‘add timeslice tocircular source buffer’ procedure 1013 then adds the timeslice audiodata to a circular buffer (or other limited-size buffer mechanism) sothat the audio information from the timeslice data is available to theother analysis threads. Finally, the ‘pre-analysis’ thread 1000 returnsto the ‘receive source packet’ procedure 1005 to receive the next audiopacket from its source.

The VAD is tuned to use one-millisecond frames to achieve the requiredtemporal resolution. Other means, such as the use of multiplepartially-overlapping frames at lower resolution, can be used to producethe required temporal resolution. The use of the VAD buffer issubsequently described with respect to FIG. 13. The VAD data is oneexample of a stream of feature data that results from analysis of theconversational communication. Other feature data can be generated byother analysis of VAD data, the audio, or other conversationalcharacteristics. The feature data can be analyzed over a communicativeinterval.

FIG. 11 illustrates a ‘user output’ thread 1100 that is invoked by the‘invoke output thread’ procedure 905 and that initiates at a ‘start’terminal 1101. The purpose of the ‘user output’ thread 1100 is toreceive modified packets (modified as described with respect to the‘adjust packet WRT output parameters’ procedure 815) from the‘distribute packet to output thread’ procedure 817 from each of thesources. The packets from each source are synchronized and mixed togenerate the audio output for the user. The audio output for one usercan be synchronized with the other outputs (so that each output sendsthe corresponding information at substantially the same time) and thenthe packet is sent to the output for presentation to the user.

After initiation, the ‘user output’ thread 1100 continues to the‘initialization’ procedure 1102 to perform any required one timeinitialization. A ‘synchronize packets’ procedure 1103 receives themodified audio packets from the ‘distribute packet to output thread’procedure 817 and synchronizes them prior to mixing. Once the packetsare synchronized, a ‘mix output packet’ procedure 1105 combines each ofthe adjusted source packets to generate an output packet. Oncegenerated, a ‘synchronize with other outputs’ procedure 1107 cansynchronize the output packet with the packets for the other outputs.Then a ‘send output packet’ procedure 1109 sends the output packet forpresentation to a user (possibly after synchronization with the otheroutputs).

FIG. 12 illustrates a ‘configuration generation’ thread 1200 that isinvoked by the ‘invoke configuration generation thread’ procedure 911,that initiates at a ‘start’ terminal 1201 and continues to an‘initialization’ procedure 1202. Next, an ‘iteration’ procedure 1203iterates each allowed configuration.

As each allowed configuration is iterated, a ‘create configurationgraph’ procedure 1205 creates a graph in memory representing the allowedconfiguration. This can include pre-weighting the edges of theconfiguration based on the weighting of the corresponding edges of theexisting participant graph. Once the graphs from all the allowedconfigurations are created, an ‘activate new configuration graphs’procedure 1207, classifies any new user(s) as a lurker (a user who hasnot sufficiently interacted with the other users to determine aconversational floor), renormalizes the probabilities to account for thenew user(s), then conditions the analysis threads to use the newconfigurations, and releases the previous graphs. The ‘configurationgeneration’ thread 1200 then completes through an ‘end’ terminal 1209.

One skilled in the art will understand that as the number ofparticipants increases, the number of possible combinations grows veryrapidly. The ‘iteration’ procedure 1203 in some embodiments can providea limitation in the number of possible configurations that are actuallyiterated.

The ‘initialization’ procedure 1202 in some configurations will onlyallow one copy of the ‘configuration generation’ thread 1200 to executeat a time because the generation of the configuration graphs can takeconsiderable computing resources. In such an embodiment, the‘initialization’ procedure 1202 remembers that a subsequent request hasbeen made while a prior request is being serviced and will condition the‘configuration generation’ thread 1200 to execute again after theconfiguration graphs from the prior request are activated.

The following is directed to the analysis performed on the audioinformation to determine the configuration probabilities used toestablish conversational group membership and to set the floor controldata structure 500.

One skilled in the art will understand that there can be considerablelatency between the time one user makes a vocalization and the time thatother users hear the vocalization. This delay is contributed to (forexample) by the processing times required to digitize the vocalization,to send the digitized vocalization to the group communication server103, the time required for the group communication server 103 to processthe vocalization, and to send it to the other users. This delay needs tocompensate for by many of the subsequently described analysisprocedures. It also means that the analysis of two users is generallynot symmetrical (the analysis of whether user A is responding to user Bis different from the analysis of whether user B is responding to user Aas in each case, the response is to what the user heard and thetimeframe of that hearing).

One aspect of the invention is that of establishment of conversationalgroup membership. “Establishment” may be expressed in a variety of waysbased on known reasoning techniques. For example, membership may imply abinary classification (“users X and Y are in conversational floor A”),or a belief such as “user X is in conversation with user Y withprobability P.”

To determine a conversational group membership, one aspect of theinvention monitors each user's vocalizations. Thus, when ‘this user’mentions another user's name at the beginning of ‘this user's’vocalization, the probability is increased that ‘this user’ and thenamed ‘other user’ are in a conversation. This is referential analysis.Furthermore, if ‘this user’ is making backchannel vocalizations(vocalizing “words” such as, but without limitation, “uh-huh”, “OK”,“yep” etc.) with respect to the vocalization of the ‘other user’, it ismore probable that ‘this user’ and the ‘other user’ are incommunication. This is responsive analysis. Finally, if ‘this user’ andthe ‘other user’ are exhibiting turn-taking behavior, it is much moreprobable that ‘this user’ and the ‘other user’ are in communication.

One aspect of the invention analyzes data for conversationalcharacteristics. These characteristics include those that can bedetermined from analysis of the audio information from a subset of theaudible sources or from physiological responses to the conversation (forexample—but without limitation, responses measured by a biometric deviceand/or information from an eye-tracker device).

Some aspects of the invention require audio phrase-recognitioncapability (that is, the ability to recognize a user's name from anaudio stream or to recognize specific short vocalizations from a user).Thus, these aspects of the invention do not require “deep” languageunderstanding but instead use word/phrase recognition technology(however, the use of “deep” language understanding could be used toimprove the analysis).

Each of the analysis modules determines quantitative measurescorresponding to the measurement of a particular “feature.” For example,one feature used by the turn-taking analysis is the amount ofoverlapping speech produced by speakers A and B over a specified timewindow.

The results of the analysis of user vocalizations or otherconversational characteristics are combined on an ongoing basis usingreasoning techniques known in the art. For example, raw quantitativefeatures can be used to compute likelihoods that are combined to computea posterior probability that speakers are or are not participating inthe same conversational floor.

The quantitative features can be computed in a pair-wise fashion; oncethe pair-wise probabilities are computed, they can then be used tocompute a conversational floor configuration that is most consistentwith all of the evidence developed from the features. One way to do thisis to divide a set of speaking users (speakers) into disjoint sets (forexample, each disjoint set containing two or more speakers). Users whohave not yet (or not recently) provided enough vocalization to be placedin a conversational floor (lurkers) can receive all communications, canbe placed in a particular conversational floor, or otherwise handled ina default manner.

One algorithm to create the sets is to enumerate all possibleconfigurations of such sets where each configuration is represented as acompletely connected weighted graph in which the nodes are users and theedge weights are the posterior probabilities that the connected usersare communicating; the configuration with the highest mean edge weightis the most likely configuration for the speakers.

Once the most probable configuration is determined (and in someembodiments after considering a lagging or smoothing effect to thechange of configuration) the set of mixing parameters for the audiomixer 301 characteristics are changed (for example, by changing thevalues in the floor control data structure 500) to enhance the outputfor each user dependent on the user's conversational floor (as has beenpreviously described with respect to FIG. 7). In one embodiment, thesechanges enhance the contribution from each of the users in the sameconversational floor as the listener while degrading the contributionfrom each of the users that are not in the same conversational floor asthe listener (other changes can be envisioned). These enhancements andderogations are specified by the values in the “mod code” field of theoutput data structure 507 and can include:

-   -   (a) Increasing the volume of the vocalizations corresponding to        members of the group and decreasing the volume of non-member        vocalizations.    -   (b) Applying conventional audio effects to “highlight” members'        vocalizations or “muddy” nonmembers' vocalizations.    -   (c) Obfuscating (partially or completely) the vocalizations        corresponding to non-members. For example, using the        block-scrambling algorithm of Smith & Hudson, Low Disturbance        Audio For Awareness and Privacy in Media Space Applications, ACM        Multimedia 95—Electronic Proceedings, Nov. 5-9, 1995.    -   (d) Suppressing all conversations except for an override        communication.    -   (e) Adding additional audio cues indicating floor membership.

One skilled in the art will understand that additional fields can beadded to the output data structure 507 to enable other types ofindications of floor membership. For textual communication, for example,the floors can be distinguished by changing the typeface, the color, thesize, shading or changing any other textual parameter. For videoindications of floor membership, the presentation of the videorepresenting the user can be adjusted such as by grouping the video ofmembers on the floor together, by changing the image size, brightness,contrast, rotation, outline, framing, frame rate, and/or by having someother graphical connections between the members of each floor (forexample, lines).

Turning now to the further discussion of some of the analysis aspects ofthe invention. FIG. 13 illustrates a ‘turn taking analysis’ thread 1300that determines when users are taking turns talking with each other.

The ‘turn taking analysis’ thread 1300 is invoked by the ‘invoketurn-taking analysis thread’ procedure 913, initiates at a ‘start’terminal 1301 and continues to an ‘initialization’ procedure 1303 thatperforms any required initialization. Then the ‘turn taking analysis’thread 1300 continues to a ‘wait for VAD update’ procedure 1305 thatdelays until at least one new entry is added to the VAD buffer by the‘add timeslice to circular source buffer’ procedure 1013. Once the newVAD entry is added, the ‘turn taking analysis’ thread 1300 continues toan ‘iterate every other source’ procedure 1307 that iterates over everysource other than the source associated with the ‘turn taking analysis’thread 1300. For each other source, an ‘align VAD times’ procedure 1308temporally aligns ‘this user's’ VAD and the ‘other user's’ VAD so thatthe subsequent feature determination uses the same time base (thussynchronizing portions of the two audio streams). Next, an ‘iterateevery feature’ procedure 1309 invokes every module that computes aquantitative value related to ‘this user’ and the ‘other user’.

For each iterated feature, a ‘determine feature from VADs’ procedure1311 examines the VADs to determine quantitative measurements ofturn-taking characteristics. Each feature can analyze an arbitraryportion of the two users' VAD buffers.

Example features include those that: (1) indicate that two speakers arein the same conversational floor (for example, but without limitation,by detecting when one speaker starts vocalizing at a transitionrelevance place (TRP) of another speaker, and by detecting when onespeaker has anticipated a TRP of another speaker, etc.); and (2)indicate that two speakers are not in the same conversational floor (forexample, but without limitation, by detecting sustained periods ofoverlapping speech with the two speakers, and by detecting a lack ofcorrelation between the speakers with respect to starting their speechat TRPs, etc.).

One skilled in the art will understand that temporally aligning the VADscan be as simple as specifying a starting bit location in the VAD bufferfor each of the VADs.

A ‘determine likelihood of shared floor’ procedure 1313 then determinesthe likelihood from the quantitative values returned from the featurethat ‘this user’ and the ‘other user’ share a conversational floor. Thelikelihoods for a feature can be determined in various ways. One suchway is by comparing the measured feature value with a distribution oflikelihoods for that feature.

After all the features are iterated, the ‘turn taking analysis’ thread1300 continues to a ‘determine single posterior probability’ procedure1315 that evaluates and combines the likelihoods and generates a singleposterior probability that ‘this user’ and the ‘other user’ are in thesame conversational floor. This can be done using a variety of methods(such as multiplying the likelihoods from statistically independentfeatures, and/or other ways known in the art to generate a posteriorprobability).

When all the ‘other users’ have been iterated, the ‘turn takinganalysis’ thread 1300 continues to a ‘send result to floor configurationthread’ procedure 1317 that sends the results of the analysis to the‘floor configuration’ thread 700 and the ‘turn taking analysis’ thread1300 returns to the ‘wait for VAD update’ procedure 1305 for the nextanalysis iteration.

The VAD buffer is large enough to account for the maximum time intervalexamined by any of the feature modules iterated by the ‘iterate everyfeature’ procedure 1309 as well as the amount of relative timeadjustment required to temporally align the ‘this user's’ VAD bufferwith the ‘other user's’ VAD buffer by the ‘align VAD times’ procedure1308. In some embodiments, the VAD buffer is large enough to store onthe order of 30 seconds of VAD information.

The amount of time required for alignment of ‘this user's’ and the‘other user's’ VAD buffer is variable, but is of the order of 350milliseconds.

Note that the pair-wise probabilities for the two users need not besymmetric. The feature computation from user 1 to user 2 is generallydifferent from the feature computation from user 2 to user 1 becauseuser 1 and user 2 VAD buffers are temporally aligned differently.

Another way to make probabilistic inferences of conversational groupmembership (that is whether two users are in the same conversationalfloor) is when one user vocalizes an identifier of another user oridentifiable group of users (for example, a proper name, nickname, callsign, group name etc.). Some of these inferences include the use ofaudible identifiers when addressing a vocalization (for example, “Jim, Ineed . . . ”), change in vocalization volume and/or other techniques. Inthis case, the principle is that a speaker tends to address certainconversational turns to specific recipients by using the recipients'proper names early in the turn. This happens frequently during the firstturn of a given schism (the schism-inducing turn). That is, when aspeaker is initiating a schism and wishes to involve another user, it iscommon for the initiating speaker to name the initial listening user.Wordspotting technology provides a sufficient capability to recognizethese names. However, full speech recognition will improve reliability.Note that the system need not know what names actually correspond towhat speakers—instead, it is sufficient to recognize that names arebeing used, since it is the use of names at the start of a vocalizationthat marks a potential schism. A wordspotting tool or alimited-vocabulary speech recognition engine can therefore be trainedusing lists of known proper names instead of the proper names of theactual speakers.

This mechanism is particularly important if one wishes to identify thefirst turn of such schisms before the schism-inducing turn ends.

Furthermore, the initial vocalization in a given sequence of turns oftalk usually has greater speech energy amplitude (volume level) than theimmediately preceding turns in the previous sequence. This can be ameans of gaining a new addressee's attention, or a way of assertingone's right to take the turn. In either case, the act of raising one'svolume is directed toward a recipient or recipients to bring theschisming act to their attention. Like the use of proper names, this isa means of identifying possible schism-inducing turns—schism-inducingturns will begin new turn sequences, though not all new turn sequencesinvolve schisms. Features are used to compute conversational floorconfigurations as described for the turn taking analysis previouslydescribed with respect to FIG. 7 and FIG. 13.

FIG. 14 illustrates a ‘referential action analysis initialization’thread 1400 that is invoked by the ‘initialize referential analysis’procedure 915 and initiates at the ‘start’ terminal 1401. The‘referential action analysis initialization’ thread 1400 then continuesto an ‘initialization’ procedure 1403 that performs any requiredinitialization. Then an ‘iterate user name variants’ procedure 1405iterates over each variant of ‘this user's’ name.

For each variant of the user's name, a ‘retrieve audio of user namevariant’ procedure 1407 retrieves audio information from storage that isthat of one variant of the user's name. This can include the given nameof the user, an identifier of a group of users, a nickname and/or ahandle.

Once the audio of the user's name variant is retrieved, a ‘trainwordspotting model for user name variant’ procedure 1409 processes theaudio data and trains the model to recognize the name variant. Dependingon the model being used, every ‘other user’ may need to provide theirown sample of ‘this user's’ name variant. Other models can be used thatwill use the name variant as spoken by ‘this user’ and allow ‘otheruser's’ use of the name to be recognized. In some embodiments, thewordspotting model is previously trained to recognize common names.

After the wordspotting model is trained, a ‘determine time-length ofuser name variant’ procedure 1411 determines the time required tovocalize the user's name variant.

After all the user's name variants have been processed, the ‘referentialaction analysis initialization’ thread 1400 continues to an ‘invokereferential analysis thread’ procedure 1413 that actually performs thereferential analysis and that is subsequently described with respect toFIG. 15. Finally, the ‘referential action analysis initialization’thread 1400 completes through an ‘end’ terminal 1415. One skilled in theart will understand that some embodiments need not train thewordspotting model as each new input is activated, but can instead (orin conjunction with) use an initially trained wordspotting model.

The ‘train wordspotting model for user name variant’ procedure 1409trains a wordspotting model for one or more forms of the ‘other user's’proper name. This can be done using techniques that apply, for example,Hidden Markov Models (HMM), and in particular HMM techniques that cantrain models dynamically. Wordspotting differs from full speechrecognition in that it only involves detecting the presence of a limitednumber of specific words (sounds) in an audio stream as opposed torecognizing all words and building a linguistic model of what is beingsaid. One skilled in the art will understand that to separate desiredwords from background sounds (those other than the desired words), somereasonably large amount of each user's speech may be required tostatistically characterize it (e.g., to create a “background HMM”). Somealgorithms require that the recording of ‘this user's’ name must bespoken by each ‘other user’ which is not unreasonable if the users usethe system frequently and use it repetitively with each other (e.g.,groups of friends who often speak to each other). In any case, morerecent advances in speaker-independent speech recognition technologiescan be applied here since the word in question is known in advance.

FIG. 15 illustrates a ‘referential action analysis’ thread 1500 that isinvoked by the ‘invoke referential analysis thread’ procedure 1413,initiates at a ‘start’ terminal 1501 and that is initialized by an‘initialization’ procedure 1503. The ‘referential action analysis’thread 1500 then continues to a ‘detect user vocalization within window’procedure 1505 that determines whether the user's vocalization is earlyin the turn (thus, the vocalization occurring where a referential wordis more likely to be used). When the user's vocalization is within thewindow, the ‘referential action analysis’ thread 1500 continues to an‘iterate every ‘other user’’ procedure 1507.

For each iterated ‘other user’, a ‘scan for ‘other user's’ name’procedure 1509 scans ‘this user's’ audio buffer using the ‘other user's’wordspotting model to determine whether ‘this user’ has vocalized a namevariant of the ‘other user’. A ‘name found’ decision procedure 1511 thendetermines whether one or more matching name variants were found. If nomatching name variant was found, the ‘referential action analysis’thread 1500 continues back to the ‘iterate every ‘other user’’ procedure1507 to check another user's name variants.

However, if a name was found at the ‘name found’ decision procedure1511, the ‘referential action analysis’ thread 1500 continues to an‘increase conversation probability’ procedure 1513 that increases theprobability that ‘this user’ is in a conversation and increases theprobability of a conversation between ‘this user’ and the ‘other user’(by adjusting the probability associated with the corresponding edgebetween ‘this user’ and the ‘other user’ in the participant graph).Then, the ‘referential action analysis’ thread 1500 continues to the‘iterate every ‘other user’’ procedure 1507 to continue examining the‘other user’ name variants.

At the completion of the ‘iterate every ‘other user’’ procedure 1507,the ‘referential action analysis’ thread 1500 continues back to the‘detect user vocalization within window’ procedure 1505.

To summarize the above, we want to know whether ‘this user’ has used‘other user's’ name at the beginning of ‘this user's’ vocalizationbecause this is evidence that ‘this user’ is trying to get ‘otheruser's’ attention. The effect is expressed as a higher probabilityrather than a binary decision because it is possible that ‘other user's’name is not actually being used (wordspotting returns a probability ofmatch) or that ‘other user's’ name is being used in some context otherthan a “hail”. Note that we limit how “far into” the vocalization welook, not so much because wordspotting algorithms are expensive tocompute (it can be done in real-time), but rather because the fartherthe word (‘other user's’ name) lies in the vocalization, the less likelyit is to be used to “hail” the ‘other user’. Furthermore, multiple namevariants can be found each with its own probability. The ‘increaseconversation probability’ procedure 1513 can pick the highestprobability match, or use some combination of matches to increase theprobability of conversation between the users.

Another way to help determine whether two users are in the sameconversational floor is to make probabilistic inferences ofconversational group membership based on one or more of ‘this user's’conversational characteristics that are responsive to acts of ‘otherusers’. In this aspect of the invention, temporal vocalization adjacencyis also relevant to the analysis. Potentially relevant user actionsinclude: Backchannel/continuers, Common content, and Prosody.

For backchannel/continuers, if the communication system provides audiocontent, then users usually produce speech (“uh huh,” “hmm,” “yeah”)that is intended primarily to affirm the current speaker's right to holdtheir turn and continue speaking. Such backchannel vocalizations, orcontinuers, can be heuristically distinguished from regular turns bytheir short length and simple intonation structure. (These measures donot require speech recognition per se. Of course, speech recognitionwill improve the ability to distinguish backchannel responses byidentifying certain vocalizations explicitly. However, heuristic“wordspotting” techniques, trained on corpora of the sounds that areknown to make up most backchannel communication, are also effective.) Ifthe communication system uses a non-synchronous medium, such backchannelinformation will be less frequent but will likely still be present.

For common content, use of similar phrasing from one turn to anotherprovides evidence that the speakers are in conversation. One way todetermine this is to compare the temporal energy profiles ofvocalizations (as in echo cancellation algorithms). Another way is tobreak vocalizations into smaller units (such as phonemes) and comparethe distributions of various n-grams of these units. Another way is touse speech recognition. For example, speech can be converted to textusing known speech recognition techniques. The text can then be comparedusing textual similarity algorithms drawn from, e.g., the informationretrieval art, with high content similarity scores being used toincrease the probability that the speakers share a conversational floor.Since turns may constitute multiple statements, it may be useful tocompare content similarity (using any of the above approaches, not justspeech recognition) at a sub-turn granularity.

Prosody refers to variation in pitch, loudness, tempo, and rhythm. Ifthe communication system provides audio content, then speakers who aresharing a conversational floor tend to produce vocalizations followingpatterns corresponding to specific prosodic profiles. For example,speakers in a shared conversational floor exhibit similarities in therhythm of their speech. That is, near-isochronous patterns of stress andemphasis can be identified for each vocalization and the rates of thesepatterns can be compared across vocalizations.

The existence of strong correlations between adjacent vocalizations bydifferent speakers is evidence of participation in the sameconversational floor. The potentially relevant user actions may beconcurrent (e.g., audible backchannel to a speaker while listening) orsequential (e.g., use of rhythm or common phrasing in subsequent speech)relative to the others' acts. Features are used to computeconversational floor configurations such as described for the turntaking analysis previously described with respect to FIG. 7 and FIG. 13.

One skilled in the art will understand that the analysis describedherein can also be applied to textual communication between more thantwo people. For example, chat, instant messaging and UNIX talk systemsenabled for more than two people. For normal chat and instant messagingsystems, where characters are not individually sent to the receivingindividuals (in contrast to multiple-party versions of UNIX talk thatindividually transmit and display each character typed) some of thetemporal window-related aspects of the analysis need not be done.However, the referential analysis directly applies as does commoncontent analysis. Furthermore, there are communication characteristicscommon to those who use chat and instant messaging system. Thesecharacteristics can be analyzed accordingly.

FIG. 16 illustrates a ‘responsive action analysis initialization’ thread1600 that is invoked by the ‘initialize responsive analysis’ procedure919, initiates at a ‘start’ terminal 1601, and continues to an‘initialization’ procedure 1603 that performs any necessaryinitialization. The ‘responsive action analysis initialization’ thread1600 then continues to an ‘iterate each backchannel word’ procedure1605. For each iteration, a ‘retrieve backchannel word audio’ procedure1607 retrieves the audio of the backchannel vocalization, and subjectsthis audio to a ‘train wordspotting model for backchannel word’procedure 1609 that trains the user's wordspotting model to recognizethe backchannel vocalization. A ‘determine time length of backchannelword’ procedure 1611 then determines the length of time required to makethe vocalization to assist the wordspotting model. After all thebackchannel words are processed, the ‘responsive action analysisinitialization’ thread 1600 continues to an ‘invoke analysis thread’procedure 1613 that invokes the analysis thread that is subsequentlydescribed with respect to FIG. 17. Finally, the ‘responsive actionanalysis initialization’ thread 1600 completes through an ‘end’ terminal1615. One skilled in the art will understand that some embodiments neednot train the wordspotting model as each new input is activated, but caninstead (or in conjunction with) use an initially trained wordspottingmodel.

FIG. 17 illustrates a ‘responsive action analysis’ thread 1700 that isinvoked by the ‘invoke analysis thread’ procedure 1613, initiates at a‘start’ terminal 1701, and initializes at an ‘initialization’ procedure1703. A ‘detect user vocalization in window’ procedure 1705 detects when‘this user’ makes a vocalization within a minimum-length window. Forthis analysis, the window is such that ‘this user’ has continuouslyvocalized for a period at least as long as the shortest time lengthcomputed by a ‘determine time length of backchannel word’ procedure1611. (If the user has not vocalized for at least this much time, noneof the wordspotting models can possibly match the current vocalization.)Further, the continuous vocalization should not contain audio samplesthat have been previously matched by a backchannel wordspotting model.Once ‘this user’ makes a vocalization within the window, the ‘responsiveaction analysis’ thread 1700 continues to an ‘iterate every other user’procedure 1707.

For each iterated user, an ‘other user speaking’ decision procedure 1709determines whether the iterated user is vocalizing within a vocalizationwindow. A user is vocalizing within the vocalization window when theuser is currently speaking (or had been recently speaking, for adefinition of “recently” that corresponds to a specified maximumperiod). If not, the ‘responsive action analysis’ thread 1700 goes backto the ‘iterate every other user’ procedure 1707.

Otherwise, the ‘responsive action analysis’ thread 1700 continues to an‘iterate user's backchannel words’ procedure 1711 that iterates each of‘this user's’ backchannel words. For each iterated backchannel word, a‘scan ‘this user's’ audio for backchannel word’ procedure 1713 scans‘this user's’ audio buffer to determine whether ‘this user's’vocalization included the backchannel word. If no match was found forthe iterated word, a ‘word found’ decision procedure 1715 causes the‘responsive action analysis’ thread 1700 to go back to the ‘iterateuser's backchannel words’ procedure 1711 to iterate the next backchannelword. However, if a match was found, the ‘responsive action analysis’thread 1700 continues to an ‘increase conversation probability’procedure 1717 that adjusts the probability in the participant graphthat ‘this user’ and the ‘other user’ are in conversation. Then the‘responsive action analysis’ thread 1700 continues to the ‘iterate everyother user’ procedure 1707 to iterate the next ‘other user’. Otherembodiments can scan all of the backchannel words and appropriatelycombine the probabilities of the found words instead of (as is done inthis implementation) advancing to the next user after the firstbackchannel word is found.

Once all the ‘other users’ have been iterated, the ‘responsive actionanalysis’ thread 1700 continues back to the ‘detect user vocalization inwindow’ procedure 1705 to detect another vocalization by ‘this user’within the minimum-length window.

FIG. 18 illustrates a set of graphs for illustration 1800. One of theillustrated graphs is a participant graph 1801 that has nodes for theusers, and weighted edges for the probability that users represented bythe nodes are in conversation. Also shown are a number of configurationgraphs that represent possible conversational configurations. Theseinclude a four-way configuration graph 1803, a first two-wayconfiguration graph 1805, a second two-way configuration graph 1807, anda third two-way configuration graph 1809.

One way to determine the most likely conversational floor configurationis to find the average of all the weights for each of theconfigurations. Thus, the four-way configuration graph 1803 has anaverage weight of 0.4, the first two-way configuration graph 1805 has anaverage weight of 0.25, the second two-way configuration graph 1807 hasan average weight of 0.1, and the third two-way configuration graph 1809has an average weight of 0.85 making it the most likely conversationalfloor configuration.

One skilled in the art will understand that the inventive techniquesdisclosed at least in FIG. 15 and FIG. 17 can be used with othercommunication besides audio communications. Thus, referential andresponsive analysis can also be applied to textual communication (suchas chat, instant messaging, or UNIX talk as well as to other types ofcommunication).

One skilled in the art will understand other implementation details thatare not germane to any particular embodiment. These details include, butare not limited to, detection of activation and deactivation of asource, any cleanup after a source is deactivated, etc.

In addition, one skilled in the art will understand that there are manyways the invention can be implemented using different architectures aswell as different embodiments of any given architecture. Thecontemplated architecture includes the range from complete hardwareimplementations through complete software implementations using anyprogramming methodology or combinations of programming methodologies andinclude the possibility of having the processing capability distributedbetween the several devices (for example, where the mixing for eachoutput is done on the device that receives the output).

Further, one skilled in the art will understand that the invention canbe augmented with additional known types of inferential analysis thatuse input data other than those directly affected by conversation (i.e.,those employed by the present invention). For example, so-calledcontext-awareness systems combine many kinds of physical sensor data andcomputer application data to make assessments of user activity. Acontext-aware system that is capable of tracking users' physicallocations within a building can compute which users are co-present in aroom; such a system might assess co-present users' vocalizations ashaving a high probability of being directed at each other as opposed tobeing directed at remote users (who are present in the computer-mediatedcommunication system but not present in the room). These other types ofinferential analysis can be integrated with the present invention in avariety of ways. For example, they can be loosely integrated in a waythat provides parameters that influence the operation of the ‘floorconfiguration’ thread 700 as previously described for manual userinputs. Alternatively, they can be tightly integrated, perhaps beingincorporated directly into a state machine that controls the operationof the present invention.

One skilled in the art will also understand that the invention allowsmultiple users of a shared communication environment to automaticallyestablish conversational floors that allow different groups of users toconverse while still having the capability of being aware of otherconversations. Automatically here means that there is no explicitcommand, control action or control word that is used to establish theconversational floor. Instead, conversational characteristics areanalyzed and used to establish the conversational floors. The inventioncan be applied to any shared environment having independently controlledoutput. Examples of such environments include audio-based Internet chatgroups, emergency response communications, telephonic conferenceconnections or other virtual telephonic party lines, teleconferencingsystems, etc.

One skilled in the art will understand that known techniques can be usedto extend the invention to shared communication environments in whichnot all users are necessarily in remote locations and associated with apersonal input device (such as a microphone) and a personal outputdevice (such as headphones). For example, rather than having users wearindividual microphones, the physical environment (such as the roomswithin a home or an office building) can be augmented with sensors thattrack individual users as well as directional beam-forming arraysconsisting of multiple microphones. In this case, the vocalizations ofeach tracked user can be captured as a separate audio stream withoutusing individually worn microphones and the invention operates asdescribed previously. As another example, a single microphone can beused to capture an audio stream corresponding to multiple users. Knownspeaker-identification techniques can be used to detect conversationalcharacteristics (such as speech activity) of multiple users even whencaptured as a single audio stream. Thus, substreams can be distinguishedfrom the single audio stream and these substreams can be mixed andanalyzed as previously described. Known audio processing algorithms canbe used to reduce the salience of particular users' vocalizations (e.g.,using subtractive “signal cancellation” techniques) in response to thesystem's floor controls. In one embodiment, the invention can be used asa form of conversationally-selective hearing aid, being applied using asingle open microphone worn by a specific user; in this case, all speechvocalizations except for those corresponding to speakers who areidentified as being in conversation with the user wearing the openmicrophone could be reduced in salience.

From the foregoing, it will be appreciated that the invention has(without limitation) the following advantages:

-   -   (a) Provides automatic self-configuration of electronically        facilitated group communications.    -   (b) Relative to computer mediated communication systems with        manual floor control: facilitates: remote group communication in        which conversational floors (sub-groups) can schism and merge        (form and coalesce) automatically. This is more lightweight        (i.e., requires less effort to use) than manual systems and        therefore more appealing to users.    -   (c) Relative to computer mediated communication systems with        manual floor control: facilitates: remote group communication in        which conversational floors can schism and merge spontaneously.        This is more natural than manual systems, in which user        interface gestures (such as mouse-clicks, button-pushes, or        voice commands directed at the computer) must be planned and        executed prior to a change in floor membership (unlike normal        conversation where it happens without prior gestures directed at        the computer), and therefore more appealing to users.    -   (d) Relative to spatial audio systems: enables the user to        distinguish audio from a primary audio conversation while        retaining the ability to attend to multiple audio conversations,        without the need to apply spatial audio effects. This obviates        the need of spatial audio systems for delivery of binaural        audio, which require (1) normal binaural hearing on the part of        the user and (2) the use of stereo speakers or headphones.    -   (e) Relative to spatial audio systems: enables straightforward        audio separation by the user of dynamic groups of multiple        speakers. Spatial audio enables separation of single speakers        (since each is mapped to a spatial location), but to support        dynamic groups of multiple speakers, spatial audio must change        the location of some speakers (e.g., by combining the speakers        into one spatial location, or moving the spatial location of the        speakers to bring them close together). If the multiple speakers        are left in their original places as conversational groups        change, the user must keep track of the various locations to        which they must attend.    -   (f) Provides a unified conceptual and implementation framework        for multi-party conversationally-responsive systems that can be        extended to different methods of determining conversational        participation, such as audio timing, audio content, text        content, and biometric measurements (each of which may be        optionally combined with explicit user actions). This allows for        sharing of code between system implementations specialized for        particular subsets of such methods as well as simplifying the        addition of new methods to existing implementations. Previous        approaches have been limited to explicit user actions.    -   (g) Provides a unified conceptual and implementation framework        for multi-party conversationally-responsive systems that can be        applied to systems using different communication delivery media,        such as audio, video and text. This allows for sharing of code        between system implementations specialized for particular        subsets of such media as well as simplifying the addition of new        media for existing implementations. Previous approaches have        been limited to products implementing a specific combination of        audio and video.

Although the present invention has been described in terms of thepresently preferred embodiments, one skilled in the art will understandthat various modifications and alterations may be made without departingfrom the scope of the invention. Accordingly, the scope of the inventionis not to be limited to the particular invention embodiments discussedherein.

1. A system for automatically adjusting floor controls for aconversation, comprising: input modules to receive audio streams, whicheach originate from an audio source; a floor analysis module to maintainfloor controls for a current configuration of at least a portion of theaudio streams, to determine conversational characteristics shared by twoor more of the audio sources, to identify possible configurations forthe audio streams based on the conversational characteristics, toperform an analysis of the current configuration and the possibleconfigurations, to apply a change threshold to the analysis, and toautomatically adjust the floor controls when the analysis satisfies thechange threshold; and an audio mixer to mix the audio streams into oneor more outputs based on the adjusted floor controls.
 2. A systemaccording to claim 1, wherein the conversational characteristics aredetermined by at least one of a turn taking analysis, a referentialaction analysis, and a responsive action analysis.
 3. A system accordingto claim 2, further comprising: a turn taking analysis module to performthe turn taking analysis by aligning the audio streams from two or moreof the sources, monitoring the aligned audio stream, and identifying theconversational characteristics from the aligned audio stream.
 4. Asystem according to claim 2, further comprising: a referential actionanalysis module to perform the referential action analysis bydetermining that at least one of the audio streams occurs during a timewindow and identifying at least one conversational characteristiccomprising one or more of a name and a name variant for at least one ofthe other audio sources during the time window.
 5. A system according toclaim 2, further comprising: a responsive action analysis module toperform the responsive action analysis by determining that at least oneof the audio streams occurs during a time window, by identifying one ormore conversational characteristics comprising a backchannel responseduring the time window, and by matching the back channel response tovocalizations of another audio stream.
 6. A system according to claim 1,wherein the conversational characteristics comprise at least one ofsustained periods of overlapping speech, a lack of correlation for abeginning of speech, speech beginning at a transition relevance place,communication identifiers, change in speech volume, speech energyamplitude, common content, prosody, and backchannel responses.
 7. Asystem according to claim 1, wherein the change threshold comprises atleast one of a minimum number of timeslices for the currentconfiguration, a minimum number of consecutive timeslices for thecurrent configuration, and a minimum net number of timeslices for thecurrent configuration.
 8. A computer-implemented method for identifyingparticipants in a conversation, comprising: receiving audio streams,which each originate from an audio source and maintaining floor controlsfor a current configuration of at least a portion of the audio streams;determining conversational characteristics shared by two or more of theaudio sources; identifying possible configurations for the audio streamsbased on the conversational characteristics; performing an analysis ofthe current configuration and the possible configurations; applying achange threshold to the analysis and automatically adjusting the floorcontrols when the analysis satisfies the change threshold; and mixingthe audio streams into one or more outputs based on the adjusted floorcontrols.
 9. A computer-implemented method according to claim 8, whereinthe conversational characteristics are determined by at least one of aturn taking analysis, a referential action analysis, and a responsiveaction analysis.
 10. A computer-implemented method according to claim 9,further comprising: performing the turn taking analysis, comprising:aligning the audio streams from two or more of the sources; monitoringthe aligned audio streams; and identifying the conversationalcharacteristics from the aligned audio stream.
 11. Acomputer-implemented method according to claim 9, further comprising:performing the referential action analysis, comprising: determining thatat least one of the audio streams occurs during a time window; andidentifying at least one conversational characteristic comprising one ormore of a name and a name variant for at least one of the other audiosources during the time window.
 12. A computer-implemented methodaccording to claim 9, further comprising: performing the responsiveaction analysis, comprising: determining that at least one of the audiostreams occurs during a time window; identifying one or moreconversational characteristics comprising a backchannel response duringthe time window; and matching the back channel response to vocalizationsof another audio stream.
 13. A computer-implemented method according toclaim 8, wherein the conversational characteristics comprise at leastone of sustained periods of overlapping speech, a lack of correlationfor a beginning of speech, speech beginning at a transition relevanceplace, communication identifiers, change in speech volume, speech energyamplitude, common content, prosody, and backchannel responses.
 14. Acomputer-implemented method according to claim 8, wherein the changethreshold comprises at least one of a minimum number of timeslices forthe current configuration, a minimum number of consecutive timeslicesfor the current configuration, and a minimum net number of timeslicesfor the current configuration.
 15. A system for identifying participantsin a conversation, comprising: input modules to receive a plurality ofaudio streams each originating from an audio source; a floor analysismodule to maintain floor controls for a current conversationalconfiguration comprising at least a portion of the audio sources, toanalyze the audio streams for conversational characteristics shared bytwo or more of the audio sources, to determine possible conversationalconfigurations of the audio sources based on the conversationalcharacteristics, to assign a probability to at least one of the possibleconversational configurations, to select one of the possibleconversational configurations based on the probability as a mostprobable conversation configuration, and to adjust the floor controlsfor the current conversational configuration according to the mostprobable conversational configuration; and mixing the audio streamsbased on the adjusted floor controls and providing the mixed audiostreams as output.
 16. A system according to claim 15, wherein theoutput provides at least one of signals capable of driving an audiosound reproduction mechanism and signals capable of electronicrecording.
 17. A system according to claim 15, wherein the floorcontrols are automatically adjusted when the current conversationalconfiguration is maintained for a minimum number of timeslices and themost probable conversational configuration is different from the currentconversational configuration.
 18. A system according to claim 15,wherein the floor controls are automatically adjusted when the currentconversational configuration is different from the most probableconversational configuration for a minimum number of consecutivetimeslices.
 19. A system according to claim 15, wherein the floorcontrols are automatically adjusted when the current conversationalconfiguration is different from the most probable conversationalconfiguration for more than a minimum net number of timeslices.
 20. Acomputer-implemented method for identifying participants in aconversation, comprising: receiving a plurality of audio streams eachoriginating from an audio source; maintaining floor controls for acurrent conversational configuration comprising at least a portion ofthe audio sources; analyzing the audio streams for conversationalcharacteristics shared by two or more of the audio sources anddetermining possible conversational configurations of the audio sourcesbased on the conversational characteristics; assigning a probability toat least one of the possible conversational configurations and selectingone of the possible conversational configurations based on theprobability as a most probable conversation configuration; adjusting thefloor controls for the current conversational configuration according tothe most probable conversational configuration; and mixing the audiostreams based on the adjusted floor controls and providing the mixedaudio streams as output.
 21. A computer-implemented method according toclaim 20, wherein the output provides at least one of signals capable ofdriving an audio sound reproduction mechanism and signals capable ofelectronic recording.
 22. A computer-implemented method according toclaim 20, wherein the floor controls are automatically adjusted when thecurrent conversational configuration is maintained for a minimum numberof timeslices and the most probable conversational configuration isdifferent from the current conversational configuration.
 23. Acomputer-implemented method according to claim 20, wherein the floorcontrols are automatically adjusted when the current conversationalconfiguration is different from the most probable conversationalconfiguration for a minimum number of consecutive timeslices.
 24. Acomputer-implemented method according to claim 20, wherein the floorcontrols are automatically adjusted when the current conversationalconfiguration is different from the most probable conversationalconfiguration for more than a minimum net number of timeslices.